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• Linux : Create a file for writing with controlled flushing to disk in large chunks [closed]

  12 août 2023, par Pete

  On Linux I have a process (ffmpeg) that writes very slowly (even slower than 1kb / s sometimes) to disk. Ffmpeg can buffer this to 256kb chunks that get written infrequently but ffmpeg hangs occasionally and if I try to detect these hangs by checking that the file is being updated I need to wait a long time between updates, up to 10 or 15 mins, otherwise I can sometimes mistakenly kill the ffmpeg process when it appears to have stopped writing when it fact its still filling its internal buffer.

  


  Theres no way to detect this it seems unless I use strace (that I can find anyway). So I am wondering about turning off buffering in ffmpeg and writing unbuffered to disk from ffmpeg.

  


  This will result in the disk constantly making tiny writes and wasting power (and probably, if I use a SSD, mess with wear levelling too).

  


  So I would like to make ffmpeg write to a 'virtual file' (in memory - either kernel memory or a process) which I can specify the flushing characteristics of. The idea being to perhaps specify flush every 2 minutes, then I can keep an eye on the file size and make sure its still being written.

  


  I don't think I've missed any other ways to do this job - even if I could watch the socket stream incoming to ffpmeg the process itself could still stop writing and lose data. Doing the buffering outside of ffmpeg seems like the best way.

  


  Is there a built in way to do this in Linux or does it mean a custom process ? I guess I know how to do this with a small C program and pipe the data in but I wonder if theres a neater way.

  


• AVCodecContext::channel_layout 0 for WAV files

  25 novembre 2020, par cmannett85

  I have been successfully loading compressed audio files using FFmpeg and querying their channel_layouts using some code I've written :

  



  AVFormatContext* fmtCxt = nullptr;
avformat_open_input( &fmtCxt, "###/440_sine.wav", nullptr, nullptr );
avformat_find_stream_info( fmtCxt, nullptr );
av_find_best_stream( fmtCxt, AVMEDIA_TYPE_AUDIO, -1, -1, nullptr, 0 );

AVCodecContext* codecCxt = fmtCxt->streams[ret]->codec;
AVCodec* codec = avcodec_find_decoder( codecCxt->codec_id );
avcodec_open2( codecCxt, codec, nullptr );

std::cout << "Channel Layout: " << codecCxt->channel_layout << std::endl;
av_dump_format( fmtCxt, 0, "###/440_sine.wav", 0 );


  



  I've removed all error checking for brevity. However for Microsoft WAV files (mono or stereo) the AVCodecContext::channel_layout member is always 0 - despite ffprobe and av_dump_format(..) both returning valid information :

  



  Input #0, wav, from '###/440_sine.wav':
Duration: 00:00:00.01, bitrate: 740 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s


  



  Also codecCxt->channels returns the correct value. Using a flac file (with exactly the same audio data generated from the same application), gives a channel_layout of 0x4 (AV_CH_FRONT_CENTER).

  


• AVCodecContext::channel_layout 0 for WAV files

  30 décembre 2013, par cmannett85

  I have been successfully loading compressed audio files using FFmpeg and querying their channel_layouts using some code I've written :

  AVFormatContext* fmtCxt = nullptr;
  
  avformat_open_input( &fmtCxt, "###/440_sine.wav", nullptr, nullptr );
  
  avformat_find_stream_info( fmtCxt, nullptr );
  
  av_find_best_stream( fmtCxt, AVMEDIA_TYPE_AUDIO, -1, -1, nullptr, 0 );
  
  
  
  AVCodecContext* codecCxt = fmtCxt->streams[ret]->codec;
  
  AVCodec* codec = avcodec_find_decoder( codecCxt->codec_id );
  
  avcodec_open2( codecCxt, codec, nullptr );
  
  
  
  std::cout << "Channel Layout: " << codecCxt->channel_layout << std::endl;
  
  av_dump_format( fmtCxt, 0, "###/440_sine.wav", 0 );

  I've removed all error checking for brevity. However for Microsoft WAV files (mono or stereo) the AVCodecContext::channel_layout member is always 0 - despite ffprobe and av_dump_format(..) both returning valid information :

  Input #0, wav, from '###/440_sine.wav':
  
  Duration: 00:00:00.01, bitrate: 740 kb/s
  
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 1 channels, s16, 705 kb/s

  Also codecCxt->channels returns the correct value. Using a flac file (with exactly the same audio data generated from the same application), gives a channel_layout of 0x4 (AV_CH_FRONT_CENTER).